blob: 8edfc6ce20120dc94317dcd5e39c2beb3c959e59 [file] [log] [blame]
* Copyright 2019 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <stdint.h>
#include "absl/types/optional.h"
#include "media/base/delayable.h"
#include "pc/jitter_buffer_delay_interface.h"
#include "rtc_base/thread.h"
namespace webrtc {
// JitterBufferDelay converts delay from seconds to milliseconds for the
// underlying media channel. It also handles cases when user sets delay before
// the start of media_channel by caching its request. Note, this class is not
// thread safe. Its thread safe version is defined in
// pc/jitter_buffer_delay_proxy.h
class JitterBufferDelay : public JitterBufferDelayInterface {
// Must be called on signaling thread.
explicit JitterBufferDelay(rtc::Thread* worker_thread);
void OnStart(cricket::Delayable* media_channel, uint32_t ssrc) override;
void OnStop() override;
void Set(absl::optional<double> delay_seconds) override;
// Throughout webrtc source, sometimes it is also called as |main_thread_|.
rtc::Thread* const signaling_thread_;
rtc::Thread* const worker_thread_;
// Media channel and ssrc together uniqely identify audio stream.
cricket::Delayable* media_channel_ = nullptr;
absl::optional<uint32_t> ssrc_;
absl::optional<double> cached_delay_seconds_;
} // namespace webrtc